Re: [asterisk-users] sipgate outgoing calls
On Thu, 19 Sep 2013, Miguel Oyarzo wrote: Challenge authentication look good. <--- SIP read from UDP:217.10.79.23:5060 ---> SIP/2.0 200 OK Are you sure this number format 01179553708 is accepted in that SIP trunk? Some VOIP providers only accept international format. when i use a softphone client to connect directly to sipgate i can dial 01179553708 and get through -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
It looks like the challenge response after INVITE is not been accepted.
Provide more detail.
$> sip set debug peer sipgate
server*CLI> sip set debug peer sipgate
SIP Debugging Enabled for IP: 217.10.79.23:5060
Really destroying SIP dialog '3ef8ff1a6ec360626af409b112b860ee@127.0.1.1'
Method: REGISTER
-- Registered SIP 'x' at 86.140.115.135 port 5060
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Executing [01179553708@default:1] Set("SIP/x-015d",
"CALLERID(num)=x") in new stack
-- Executing [01179553708@default:2] Dial("SIP/x-015d",
"SIP/01179553708@sipgate,30,trg") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Called 01179553708@sipgate
[Sep 19 10:50:15] NOTICE[28232]: chan_sip.c:17885 handle_response_invite:
Failed to authenticate on INVITE to '"x"
;tag=as629ee6f8'
-- SIP/sipgate-015e is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [01179553708@default:3]
Hangup("SIP/x-015d", "") in new stack
== Spawn extension (default, 01179553708, 3) exited non-zero on
'SIP/x-015d'
---
server*CLI>
<--- SIP read from UDP:217.10.79.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK73a1638c;rport=5060
From: "asterisk" ;tag=as5dcb32d8
To: ;tag=99199803810c7e807ea44745826d9aa4.df2d
Call-ID: 65cb07675eefaaef5f655e8a0be6b2f6@92.63.131.3
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0
<->
--- (11 headers 0 lines) ---
Really destroying SIP dialog
'65cb07675eefaaef5f655e8a0be6b2f6@92.63.131.3' Method: OPTIONS
[Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:11588 sip_reregister:--
Re-registration for xxx...@sipgate.co.uk
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 217.10.79.23:5060:
REGISTER sip:sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK13b202d7;rport
Max-Forwards: 70
From: ;tag=as19513575
To:
Call-ID: 3ef8ff1a6ec360626af409b112b860ee@127.0.1.1
CSeq: 182 REGISTER
User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4
Authorization: Digest username="xx", realm="sipgate.co.uk",
algorithm=MD5, uri="sip:sipgate.co.uk",
nonce="523ac9531b1cc7962e07bce6a76683ee24da44d0",
response="c82fac231a41085c275899ad84f73317"
Expires: 120
Contact:
Content-Length: 0
---
server*CLI>
<--- SIP read from UDP:217.10.79.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
92.63.131.3:5060;received=92.63.131.3;branch=z9hG4bK13b202d7;rport=5060
From: ;tag=as19513575
To: ;tag=c3e497ecaece77a8e244e564b4212178.3e46
Call-ID: 3ef8ff1a6ec360626af409b112b860ee@127.0.1.1
CSeq: 182 REGISTER
Contact: ;expires=120
Content-Length: 0
<->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog
'3ef8ff1a6ec360626af409b112b860ee@127.0.1.1' in 32000 ms (Method:
REGISTER)
[Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:18301
handle_response_register: Outbound Registration: Expiry for sipgate.co.uk
is 120 sec (Scheduling reregistration in 105 s)
Reliably Transmitting (no NAT) to 217.10.79.23:5060:
OPTIONS sip:sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK2d73294a;rport
Max-Forwards: 70
From: "asterisk" ;tag=as5afd24b2
To:
Contact:
Call-ID: 1becd4dc336869c4692fc4e55e109562@92.63.131.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4
Date: Thu, 19 Sep 2013 09:51:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
server*CLI>
<--- SIP read from UDP:217.10.79.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK2d73294a;rport=5060
From: "asterisk" ;tag=as5afd24b2
To: ;tag=99199803810c7e807ea44745826d9aa4.c753
Call-ID: 1becd4dc336869c4692fc4e55e109562@92.63.131.3
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0
<->
--- (11 headers 0 lines) ---
Really destroying SIP dialog
'1becd4dc336869c4692fc4e55e109562@92.63.131.3' Method: OPTIONS
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
On Thu, 19 Sep 2013, David Duffett wrote:
i am getting these errors in asterisk cli
-- Executing [01179553708@default:1] Set("SIP/-015b",
"CALLERID(num)=xx") in new stack
-- Executing [01179553708@default:2] Dial("SIP/-015b",
"SIP/01179553708@sipgate,30,trg") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Called 01179553708@sipgate
[Sep 19 10:08:03] NOTICE[28232]: chan_sip.c:17885 handle_response_invite:
Failed to authenticate on INVITE to '""
;tag=as055d9532'
-- SIP/sipgate-015c is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
any further ideas ?
many thanks
I believe registration is in place, otherwise inbound calls would not work.
Also, registration is not required for outbound calls to work.
I would suggest cutting down your sip.conf profile to this minimal
configuration:
host=sipgate.co.uk
username=xxx
fromuser=xxx
insecure=invite,port
secret=xxx
context=my-inbound-context
type=peer
If outbound calls still do not with this, I would suggest that there may be
an issue in the general section of your sip.conf
Assuming calls do work, you can then add any other configuration lines you
feel are necessary - but remember, as with all Asterisk configuration files,
less is more :-)
On 18 Sep 2013 22:06, "Administrator TOOTAI" wrote:
Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit :
Hello
Hi
i am trying to setup sipgate gateway
i can get incoming calls fine, but when i dial in and
then try to dial
out i get this in asterisk command line
-- Called 01179248615@sipgate
[Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
handle_response_invite: Failed to authenticate on
INVITE to
'"01179553708"
;tag=as30eb9dd1'
-- SIP/sipgate-014d is circuit-busy
== Everyone is busy/congested at this time
(1:0/1/0)
here is my sip.conf file
[general]
port = 5060
bindaddr = 0.0.0.0
context=default
qualify=no
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
videosupport=yes
alwaysauthreject=yes
register => SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID
[sipgate]
type=peer
secret=SIP_PASSWORD
insecure=invite
username=SIP-ID
defaultuser=SIP-ID
fromuser=SIP-ID
context=sipgate_in
fromdomain=sipgate.co.uk
host=sipgate.co.uk
outboundproxy=proxy.live.sipgate.co.uk
qualify=yes
disallow=all
allow=alaw
dtmfmode=rfc2833
SIP-ID:SIP-Password
obviously, i replace these with my login details
but, are these the same thing ?
SIP-Password
SIP_PASSWORD
the sipgate guides are contradictory
http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk
http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri
sk
any suggestions ?
Many thanks
My setup with sipgate.de
[sipgate]
type=peer
secret=MY-PASSWORD
defaultuser=SIP-ID
host=217.10.79.9
fromuser=SIP-ID
fromdomain=sipgate.de
context=incoming-sipgate
;qualify=900
dtmfmode=info
directmedia=yes
insecure=port,invite
disallow=all
allow=ulaw,alaw
accountcode=MY-ACCOUNTCODE
What you forget is to register with them:
; Sipgate
register => SIP-ID:my-passw...@sipgate.de/SIP-ID ;don't accept to
register without FQDN
Hope that help
--
Daniel
--
_
-- Bandwidth and Colocation Provided by
http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sipgate outgoing calls
Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615@sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" ;tag=as30eb9dd1' -- SIP/sipgate-014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register => SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SIP_PASSWORD the sipgate guides are contradictory http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri sk any suggestions ? Many thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and encryption
i am running Asterisk 1.6.2.5-0ubuntu1.4 and would like to know how to incorporate [default] encryption can you point me to any guides please ? do i need to upgrade ? many thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
